3 Commits

Author	SHA1	Message	Date
Jonas Olsson	a4d873786f	Format almost everything. ... This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}	2019-07-08 13:45:15 +00:00
Benjamin Wright	150a907403	FrameEncryption Video End To End Testcase. ... There was a suggestion in a previous CL to add an end to end test case to prevent future regressions. I have enabled this by adding two fakes that perform fake encryption and enabling an end to end test with VP8 and the GenericDescriptor. Bug: webrtc:9927 Change-Id: Icf96eeed541ada1e0579eb81b6f87a46d1c43d96 Reviewed-on: https://webrtc-review.googlesource.com/c/108020 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25399}	2018-10-26 23:19:31 +00:00
Benjamin Wright	84583f6183	Enable End-to-End Encrypted Audio Payloads. ... This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender then each outgoing audio payload will first pass through the provided FrameEncryptor which will have a chance to modify the payload contents for the purposes of encryption. If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming audio payload will first pass through the provided FrameDecryptor which have a chance to modify the payload contents for the purpose of decryption. While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not use it and so it is left as null. Bug: webrtc:9681 Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf Reviewed-on: https://webrtc-review.googlesource.com/c/101702 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25001}	2018-10-04 22:08:34 +00:00